/*
    SDL - Simple DirectMedia Layer
    Copyright (C) 1997-2009 Sam Lantinga

    This library is free software; you can redistribute it and/or
    modify it under the terms of the GNU Lesser General Public
    License as published by the Free Software Foundation; either
    version 2.1 of the License, or (at your option) any later version.

    This library is distributed in the hope that it will be useful,
    but WITHOUT ANY WARRANTY; without even the implied warranty of
    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
    Lesser General Public License for more details.

    You should have received a copy of the GNU Lesser General Public
    License along with this library; if not, write to the Free Software
    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA

    Sam Lantinga
    slouken@libsdl.org
*/

/**
 *  @file SDL_audio.h
 *  Access to the raw audio mixing buffer for the SDL library
 */

#ifndef _SDL_audio_h
#define _SDL_audio_h

#include "SDL_stdinc.h"
#include "SDL_error.h"
#include "SDL_endian.h"
#include "SDL_mutex.h"
#include "SDL_thread.h"
#include "SDL_rwops.h"

#include "begin_code.h"
/* Set up for C function definitions, even when using C++ */
#ifdef __cplusplus
extern "C" {
#endif

    /**
     * When filling in the desired audio spec structure,
     * - 'desired->freq' should be the desired audio frequency in samples-per-second.
     * - 'desired->format' should be the desired audio format.
     * - 'desired->samples' is the desired size of the audio buffer, in samples.
     *     This number should be a power of two, and may be adjusted by the audio
     *     driver to a value more suitable for the hardware.  Good values seem to
     *     range between 512 and 8096 inclusive, depending on the application and
     *     CPU speed.  Smaller values yield faster response time, but can lead
     *     to underflow if the application is doing heavy processing and cannot
     *     fill the audio buffer in time.  A stereo sample consists of both right
     *     and left channels in LR ordering.
     *     Note that the number of samples is directly related to time by the
     *     following formula:  ms = (samples*1000)/freq
     * - 'desired->size' is the size in bytes of the audio buffer, and is
     *     calculated by SDL_OpenAudio().
     * - 'desired->silence' is the value used to set the buffer to silence,
     *     and is calculated by SDL_OpenAudio().
     * - 'desired->callback' should be set to a function that will be called
     *     when the audio device is ready for more data.  It is passed a pointer
     *     to the audio buffer, and the length in bytes of the audio buffer.
     *     This function usually runs in a separate thread, and so you should
     *     protect data structures that it accesses by calling SDL_LockAudio()
     *     and SDL_UnlockAudio() in your code.
     * - 'desired->userdata' is passed as the first parameter to your callback
     *     function.
     *
     * @note The calculated values in this structure are calculated by SDL_OpenAudio()
     *
     */
    typedef struct SDL_AudioSpec {
        int freq;       /**< DSP frequency -- samples per second */
        Uint16 format;      /**< Audio data format */
        Uint8  channels;    /**< Number of channels: 1 mono, 2 stereo */
        Uint8  silence;     /**< Audio buffer silence value (calculated) */
        Uint16 samples;     /**< Audio buffer size in samples (power of 2) */
        Uint16 padding;     /**< Necessary for some compile environments */
        Uint32 size;        /**< Audio buffer size in bytes (calculated) */
        /**
         *  This function is called when the audio device needs more data.
         *
         *  @param[out] stream  A pointer to the audio data buffer
         *  @param[in]  len The length of the audio buffer in bytes.
         *
         *  Once the callback returns, the buffer will no longer be valid.
         *  Stereo samples are stored in a LRLRLR ordering.
         */
        void (SDLCALL *callback)(void *userdata, Uint8 *stream, int len);
        void  *userdata;
    } SDL_AudioSpec;

    /**
     *  @name Audio format flags
     *  defaults to LSB byte order
     */
    /*@{*/
#define AUDIO_U8    0x0008  /**< Unsigned 8-bit samples */
#define AUDIO_S8    0x8008  /**< Signed 8-bit samples */
#define AUDIO_U16LSB    0x0010  /**< Unsigned 16-bit samples */
#define AUDIO_S16LSB    0x8010  /**< Signed 16-bit samples */
#define AUDIO_U16MSB    0x1010  /**< As above, but big-endian byte order */
#define AUDIO_S16MSB    0x9010  /**< As above, but big-endian byte order */
#define AUDIO_U16   AUDIO_U16LSB
#define AUDIO_S16   AUDIO_S16LSB

    /**
     *  @name Native audio byte ordering
     */
    /*@{*/
#if SDL_BYTEORDER == SDL_LIL_ENDIAN
#define AUDIO_U16SYS    AUDIO_U16LSB
#define AUDIO_S16SYS    AUDIO_S16LSB
#else
#define AUDIO_U16SYS    AUDIO_U16MSB
#define AUDIO_S16SYS    AUDIO_S16MSB
#endif
    /*@}*/

    /*@}*/


    /** A structure to hold a set of audio conversion filters and buffers */
    typedef struct SDL_AudioCVT {
        int needed;         /**< Set to 1 if conversion possible */
        Uint16 src_format;      /**< Source audio format */
        Uint16 dst_format;      /**< Target audio format */
        double rate_incr;       /**< Rate conversion increment */
        Uint8 *buf;         /**< Buffer to hold entire audio data */
        int    len;         /**< Length of original audio buffer */
        int    len_cvt;         /**< Length of converted audio buffer */
        int    len_mult;        /**< buffer must be len*len_mult big */
        double len_ratio;   /**< Given len, final size is len*len_ratio */
        void (SDLCALL *filters[10])(struct SDL_AudioCVT *cvt, Uint16 format);
        int filter_index;       /**< Current audio conversion function */
    } SDL_AudioCVT;


    /* Function prototypes */

    /**
     * @name Audio Init and Quit
     * These functions are used internally, and should not be used unless you
     * have a specific need to specify the audio driver you want to use.
     * You should normally use SDL_Init() or SDL_InitSubSystem().
     */
    /*@{*/
    extern DECLSPEC int SDLCALL SDL_AudioInit(const char *driver_name);
    extern DECLSPEC void SDLCALL SDL_AudioQuit(void);
    /*@}*/

    /**
     * This function fills the given character buffer with the name of the
     * current audio driver, and returns a pointer to it if the audio driver has
     * been initialized.  It returns NULL if no driver has been initialized.
     */
    extern DECLSPEC char *SDLCALL SDL_AudioDriverName(char *namebuf, int maxlen);

    /**
     * This function opens the audio device with the desired parameters, and
     * returns 0 if successful, placing the actual hardware parameters in the
     * structure pointed to by 'obtained'.  If 'obtained' is NULL, the audio
     * data passed to the callback function will be guaranteed to be in the
     * requested format, and will be automatically converted to the hardware
     * audio format if necessary.  This function returns -1 if it failed
     * to open the audio device, or couldn't set up the audio thread.
     *
     * The audio device starts out playing silence when it's opened, and should
     * be enabled for playing by calling SDL_PauseAudio(0) when you are ready
     * for your audio callback function to be called.  Since the audio driver
     * may modify the requested size of the audio buffer, you should allocate
     * any local mixing buffers after you open the audio device.
     *
     * @sa SDL_AudioSpec
     */
    extern DECLSPEC int SDLCALL SDL_OpenAudio(SDL_AudioSpec *desired, SDL_AudioSpec *obtained);

    typedef enum {
        SDL_AUDIO_STOPPED = 0,
        SDL_AUDIO_PLAYING,
        SDL_AUDIO_PAUSED
    } SDL_audiostatus;

    /** Get the current audio state */
    extern DECLSPEC SDL_audiostatus SDLCALL SDL_GetAudioStatus(void);

    /**
     * This function pauses and unpauses the audio callback processing.
     * It should be called with a parameter of 0 after opening the audio
     * device to start playing sound.  This is so you can safely initialize
     * data for your callback function after opening the audio device.
     * Silence will be written to the audio device during the pause.
     */
    extern DECLSPEC void SDLCALL SDL_PauseAudio(int pause_on);

    /**
     * This function loads a WAVE from the data source, automatically freeing
     * that source if 'freesrc' is non-zero.  For example, to load a WAVE file,
     * you could do:
     *  @code SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...); @endcode
     *
     * If this function succeeds, it returns the given SDL_AudioSpec,
     * filled with the audio data format of the wave data, and sets
     * 'audio_buf' to a malloc()'d buffer containing the audio data,
     * and sets 'audio_len' to the length of that audio buffer, in bytes.
     * You need to free the audio buffer with SDL_FreeWAV() when you are
     * done with it.
     *
     * This function returns NULL and sets the SDL error message if the
     * wave file cannot be opened, uses an unknown data format, or is
     * corrupt.  Currently raw and MS-ADPCM WAVE files are supported.
     */
    extern DECLSPEC SDL_AudioSpec *SDLCALL SDL_LoadWAV_RW(SDL_RWops *src, int freesrc, SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len);

    /** Compatibility convenience function -- loads a WAV from a file */
#define SDL_LoadWAV(file, spec, audio_buf, audio_len) \
    SDL_LoadWAV_RW(SDL_RWFromFile(file, "rb"),1, spec,audio_buf,audio_len)

    /**
     * This function frees data previously allocated with SDL_LoadWAV_RW()
     */
    extern DECLSPEC void SDLCALL SDL_FreeWAV(Uint8 *audio_buf);

    /**
     * This function takes a source format and rate and a destination format
     * and rate, and initializes the 'cvt' structure with information needed
     * by SDL_ConvertAudio() to convert a buffer of audio data from one format
     * to the other.
     *
     * @return This function returns 0, or -1 if there was an error.
     */
    extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT *cvt,
                                                  Uint16 src_format, Uint8 src_channels, int src_rate,
                                                  Uint16 dst_format, Uint8 dst_channels, int dst_rate);

    /**
     * Once you have initialized the 'cvt' structure using SDL_BuildAudioCVT(),
     * created an audio buffer cvt->buf, and filled it with cvt->len bytes of
     * audio data in the source format, this function will convert it in-place
     * to the desired format.
     * The data conversion may expand the size of the audio data, so the buffer
     * cvt->buf should be allocated after the cvt structure is initialized by
     * SDL_BuildAudioCVT(), and should be cvt->len*cvt->len_mult bytes long.
     */
    extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT *cvt);


#define SDL_MIX_MAXVOLUME 128
    /**
     * This takes two audio buffers of the playing audio format and mixes
     * them, performing addition, volume adjustment, and overflow clipping.
     * The volume ranges from 0 - 128, and should be set to SDL_MIX_MAXVOLUME
     * for full audio volume.  Note this does not change hardware volume.
     * This is provided for convenience -- you can mix your own audio data.
     */
    extern DECLSPEC void SDLCALL SDL_MixAudio(Uint8 *dst, const Uint8 *src, Uint32 len, int volume);

    /**
     * @name Audio Locks
     * The lock manipulated by these functions protects the callback function.
     * During a LockAudio/UnlockAudio pair, you can be guaranteed that the
     * callback function is not running.  Do not call these from the callback
     * function or you will cause deadlock.
     */
    /*@{*/
    extern DECLSPEC void SDLCALL SDL_LockAudio(void);
    extern DECLSPEC void SDLCALL SDL_UnlockAudio(void);
    /*@}*/

    /**
     * This function shuts down audio processing and closes the audio device.
     */
    extern DECLSPEC void SDLCALL SDL_CloseAudio(void);


    /* Ends C function definitions when using C++ */
#ifdef __cplusplus
}
#endif
#include "close_code.h"

#endif /* _SDL_audio_h */
